| 00:02.09 | *** join/#asterisk SaiSoma (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 00:02.30 | p3nguin | igramul: "core show application MeetMe" |
| 00:03.18 | *** join/#asterisk jks (~jks@193.189.93.254) |
| 00:03.29 | manxpower | Channel banks are not mentioned in The Book? |
| 00:07.29 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 00:08.34 | Katty | hi |
| 00:08.42 | Katty | i just got back from a 2 mile walk with the pup |
| 00:10.37 | jaytee | yay |
| 00:20.13 | *** join/#asterisk |amadeus| (~amadeus@75.130.147.152) |
| 00:25.58 | |amadeus| | hi, is there any chance to stream an actual phone call via early media ('musiconhold') without touching source code? |
| 00:25.59 | *** join/#asterisk overfien (~overfien@c-98-247-121-77.hsd1.wa.comcast.net) |
| 00:35.35 | ruben23 | hi i have E1 leased line up to 10 Mbps used for data and voice----> is this a BRI or PRI connection..? i got no idea about it..i used it for voip. |
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| 00:58.28 | killfill | hi |
| 00:58.53 | killfill | where can i see the meaning of the status field of the ExtensionStatus AMI event? |
| 00:59.10 | killfill | i see '1'.. what is 1?.. :P |
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| 01:04.33 | jaytee | it's a numeral |
| 01:11.38 | jaytee | kill |
| 01:11.55 | jaytee | killfill, 1 indicates In Use |
| 01:12.04 | jaytee | 2 indicates Busy |
| 01:12.09 | jaytee | 0 is idle |
| 01:12.23 | jaytee | -1 indicates a hint was not found |
| 01:12.42 | jaytee | and -1 indicates the extension is removed |
| 01:12.56 | jaytee | oops, -2 is extension removed |
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| 01:13.15 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 01:33.36 | sawgood | Can someone plesae tell me what directory my IVR .mp3 files should be put in? I have to rebuild an IVR process (all I have is the files not the IVR syntax) |
| 01:34.03 | sawgood | I'll rebuild the IVR menu options, but I need to know where to put the sound files, so the IVR will know where to find them |
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| 01:40.17 | TJNII | Aah, clean keyboard. |
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| 01:55.20 | antonioclj | Hello people. This channel is a official digium support ? |
| 01:56.30 | antonioclj | somebody from digium here ? |
| 01:56.34 | AeroCloud | not that I know of, I think its just a bunch of people who try to help with asterisk |
| 01:56.55 | AeroCloud | just ask a question and if someone knows they will help you |
| 01:57.18 | AeroCloud | sometimes takes a few hours for a response, sometimes you have to ask multiple times |
| 01:58.01 | antonioclj | ok, so this channel is not a official support from digium, ok ? |
| 01:58.09 | AeroCloud | thats what I think ^ |
| 01:58.20 | antonioclj | ok, thanks. |
| 01:58.31 | AeroCloud | why are you wondering? |
| 01:59.05 | antonioclj | i purchased two g729 but don't receive any instructions about download |
| 01:59.19 | antonioclj | i'm testing a asterisk with this codec |
| 01:59.39 | AeroCloud | yeah noone can help you with that here |
| 01:59.56 | antonioclj | ok, thanks again |
| 02:00.11 | AeroCloud | np, try calling them or sending them an email |
| 02:00.28 | antonioclj | yes, i will make this. |
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| 03:42.49 | SouthSuburbTech | i need to set up and config iax2 |
| 03:43.50 | p3nguin | Are you ever going to let us know what we're supposed to do about it? |
| 03:44.06 | p3nguin | You've mentioned it a few times, but never told us what we are supposed to do. |
| 03:46.32 | SouthSuburbTech | i have set up the extension in asterisk for IAX2 with The Exten Of 200, now On Another Machine I try to connect, it attempts to register, but get stuck doing so |
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| 03:49.39 | SouthSuburbTech | help |
| 03:49.59 | SouthSuburbTech | can you assist and helping me find whats wrong |
| 03:54.01 | SouthSuburbTech | >:o |
| 03:54.10 | SouthSuburbTech | for the love of god |
| 04:00.13 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 04:07.07 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 04:08.27 | sawgood | I had to rebuild an Asterisk 1.4.21.x box (and now it is running 1.6.0.26) ... the old box had a solid IVR which played using .mp3 files from /var/lib/asterisk/sounds ... but when I rebuild the box and put the .mp3 files in the same directory ... I only get dead air when I call ... |
| 04:08.41 | sawgood | Any tips to make this IVR play would be greatly appreciated |
| 04:09.22 | SouthSuburbTech | COULD SOMEONE PLEASE HELP ME SET UP AND CONFIGURE IAX2 |
| 04:10.46 | *** join/#asterisk krynnotaur (~krynnotau@unaffiliated/krynnotaur) |
| 04:19.37 | sawgood | figured it out, I have the IVR files playing just fine now |
| 04:23.45 | p3nguin | Has no one ever given that guy the ~ask or ~answers thingies? |
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| 04:58.10 | pentanol | hi |
| 04:58.17 | pentanol | anybody alive? |
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| 05:24.50 | ChannelZ | nope |
| 05:26.23 | hardwire | just us zombies |
| 05:26.36 | hardwire | ChannelZ: you vamp or zombie? |
| 05:26.58 | ChannelZ | Neither, I'm the one with guns |
| 05:27.26 | hardwire | double tap |
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| 05:27.41 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 05:27.55 | coppice | some people are busy having lunch |
| 05:28.28 | ChannelZ | or midnight snacks as the case may be |
| 05:33.40 | hardwire | mmm.. brainznackz. |
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| 05:39.18 | pentanol | I need suggestion would it work or not.... |
| 05:39.42 | pentanol | well, I've balanced opensips based and few target asterisk nodes |
| 05:41.41 | pentanol | when client invates in same meeting room and balanced between different asterisk they not hears subscribers from another asterisk node |
| 05:44.52 | ChannelZ | How are they connecting together? Does one side call an extension on the other that dumps them into the same conference? |
| 05:45.37 | pentanol | yes indeed |
| 05:45.51 | pentanol | client calls same did |
| 05:46.19 | pentanol | i.e its't call routing issue |
| 05:46.39 | pentanol | because client reach them rooms well |
| 05:48.00 | pentanol | perhaps one asterisk don't know about another and can't broadcasting rtp traffic there |
| 05:51.55 | ChannelZ | But I mean.. lets call them Server A and Server B. Server A is hosting the conference. How do the people on Server B dial into the conference? They dial an extension on Server B that calls Server A with the extension to the same conference? How is that connection made, SIP or IAX2? |
| 05:56.52 | pentanol | not sure what you mean, but I didi in by this tutorial and add here balancing i.e. if I lose one asterisk it continue working with available http://www.opensips.org/Resources/DocsTutAsterisk |
| 05:59.21 | *** join/#asterisk sN|Connor (~connor@75.133.168.106) |
| 05:59.50 | sN|Connor | Is there anyone here that is available to help me decide if Asterisk is the right solution for me? |
| 06:01.20 | ChannelZ | ~ask |
| 06:01.21 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 06:04.09 | sN|Connor | I am running CentOS4 and have no clue really what I am doing with VoIP other than the fact this is better to have since I have staff members all across the globe. I am looking for a solution that can co-exist with my servers that I currently have and I need it to run alongside my current apps, I am thinking about putting it on my web server and I also don't know where to get a phone number from or anything like that so any help would be appriciated |
| 06:04.47 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
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| 06:06.23 | ChannelZ | Well yes you can run it along side other things but it's somewhat recommended not to depending on the amount of traffic/etc. There are lots of ITSPs (VoIP providers) that can do the actual phone line termination. |
| 06:07.34 | sN|Connor | Do you have any VoIP providers that you would recommend? |
| 06:08.00 | ChannelZ | ~itsplist-us |
| 06:08.01 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 06:08.15 | ChannelZ | I run analog phone hardware so I've never used any ITSPs personally |
| 06:09.20 | ChannelZ | You might need multiple ITSPs that have termination in whatever countries you want to make/receive phone calls in |
| 06:10.11 | pentanol | ChannelZ so what you thought about balanced clients... |
| 06:10.45 | sN|Connor | What about clients, is there any preferred "free" clients? |
| 06:11.02 | ChannelZ | pentanol: dunno that's way too much stuff for me to read and understand to make a guess at what is wrong here |
| 06:11.11 | ChannelZ | sN|Connor: Free clients for what? |
| 06:12.54 | ChannelZ | sN|Connor: Are you wanting to build a private PBX (you said you have 'clients all over the globe') so they can talk to each other, or are they needing to call lots of external phone numbers? |
| 06:15.41 | sN|Connor | It will be both. |
| 06:16.31 | sN|Connor | Im trying the yum install right now, and Im not having any luck |
| 06:17.42 | ChannelZ | how so |
| 06:19.08 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 06:20.14 | sN|Connor | Parsing package install arguments |
| 06:20.14 | sN|Connor | No Match for argument: asterisk16 |
| 06:20.14 | sN|Connor | No Match for argument: asterisk16-configs |
| 06:20.14 | sN|Connor | No Match for argument: asterisk16-voicemail |
| 06:20.14 | sN|Connor | No Match for argument: dahdi-linux |
| 06:20.14 | sN|Connor | No Match for argument: dahd |
| 06:20.15 | sN|Connor | Nothing to do |
| 06:20.15 | sN|Connor | <PROTECTED> |
| 06:21.02 | sN|Connor | I made the repo files as stated in the instructions |
| 06:24.06 | ChannelZ | hmm looks like there is nothing actually there |
| 06:24.31 | ChannelZ | for centos4 anyway |
| 06:24.50 | ChannelZ | just build from the source |
| 06:24.50 | manxpower | Packages are not supported here. |
| 06:26.03 | sN|Connor | is there a prefered place to download the source? |
| 06:26.22 | ChannelZ | asterisk.org |
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| 06:33.32 | devoid | hehe |
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| 10:02.48 | EmleyMoor | Is there a reasonably easy way to add ilbc to a Debian-packaged asterisk 1.4? |
| 10:10.39 | Tim_Toady | EmleyMoor download theh debian source add the ilbc patch and rebuild the deb packages |
| 10:11.14 | Tim_Toady | aptitude build-dep asterisk will fetch the deps for compiling asterisk |
| 10:13.02 | Tim_Toady | apt-get source asterisk will fetach the asterisk source |
| 10:13.03 | EmleyMoor | Tim_Toady: I'll see if it works - I do my own build anyway due to a patch I need |
| 10:14.02 | Tim_Toady | ok then, juts co to contrib/scripts in ur source tree and run get_ilbc_source.sh |
| 10:14.44 | Tim_Toady | and copy the source files to codecs/ilbc |
| 10:15.40 | EmleyMoor | Ah - that's what I missed |
| 10:16.37 | EmleyMoor | Hmmm... looks like they are already there from my earlier attempt 18 months ago |
| 10:16.42 | coppice | why would a rational person want to add iLBC support to anything? |
| 10:17.06 | EmleyMoor | Obviously there is something I need to do to make it happen |
| 10:17.16 | EmleyMoor | coppice: Who said I was rational? <g> |
| 10:17.34 | EmleyMoor | coppice: Actually, it may help when connecting over 3G |
| 10:20.42 | EmleyMoor | Hmmm... still looking into my previous seemingly half-done attempt |
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| 11:09.16 | |amadeus| | hi, is there any chance to stream an actual phone call via early media ('musiconhold') without touching source code? |
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| 12:16.26 | davix | anyone has a working method for streaming asx on music-on-hold? |
| 12:23.24 | |amadeus| | mms://? |
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| 12:34.21 | Meaw | hi |
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| 12:39.44 | Meaw | I've 2 sangoma cards in one box.. configured first one working fine |
| 12:40.05 | Meaw | where should I add the 2rd card configurations? in the same file wanpipe1.conf? |
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| 12:46.47 | Meaw | any thoughts? |
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| 13:27.08 | resno | hey. i am thinking about setting up asterisk, i have found some guides. i am trrying to gauge what i should expect before buying an unlocked spa-3000 |
| 13:30.53 | *** join/#asterisk hudony (~chatzilla@modemcable216.175-81-70.mc.videotron.ca) |
| 13:31.07 | hudony | Hi, I need help with multiple extension rigning |
| 13:31.33 | hudony | For some reasons, I can only answer from 1 of the 3 phone rigning |
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| 13:36.43 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 13:50.29 | sawgood | Is there a need to change the default password for anything ARI related in Asterisk 1.6.0.26? |
| 13:52.48 | [TK]D-Fender | sawgood: wrong channel <- |
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| 13:56.24 | resno | hey. i am thinking about setting up asterisk, i have found some guides. i am trrying to gauge what i should expect before buying an unlocked spa-3000 |
| 13:57.10 | *** join/#asterisk sourcode (~code@ppp-58-11-76-52.revip2.asianet.co.th) |
| 13:57.28 | [TK]D-Fender | resno: SPA-3000 is a long discontinued model |
| 13:57.35 | [TK]D-Fender | resno: What are your actual goals? |
| 13:57.45 | resno | my goal is to create a home phone system |
| 13:57.53 | killfill | hello |
| 13:57.57 | [TK]D-Fender | resno: Connecting to what? What kind of deployment? |
| 13:58.08 | resno | to a ubuntu linux server, that i have |
| 13:58.36 | killfill | i have two extensions one in Zap and one in SIP. Im calling each other, and noticed AMI is triggering ExtensionStatus events for the Zap phone. For the Sip it does not do it. |
| 13:58.47 | resno | [TK]D-Fender: to a ubuntu linux server, that i have |
| 13:58.57 | killfill | is this normal? i mean.. do i need to do a special configuration to get ExtensionStatus events of the SIP phone? |
| 13:59.02 | killfill | or its not possible |
| 13:59.12 | [TK]D-Fender | resno: ... what kind of lines? How many? What kind of phones? How many? |
| 13:59.31 | *** join/#asterisk Pan3D (~Pan3D@streaming.sensoryresearch.net) |
| 13:59.50 | resno | [TK]D-Fender: at this moment, only one line. 1 phone. hopefully analog phones |
| 14:00.15 | [TK]D-Fender | resno: Then go for a Linksys SPA-3102. It is current. |
| 14:00.32 | [TK]D-Fender | killfill: Are you referring to AMI for a single call with 2 legs? |
| 14:00.51 | resno | [TK]D-Fender: what should i expect as far as configuration and maintance? |
| 14:01.24 | killfill | [TK]D-Fender: im listening to ami events and see what i get when calling |
| 14:01.39 | [TK]D-Fender | resno: pretty small. You're line config will look almost the same as your phone port config on it. |
| 14:02.51 | [TK]D-Fender | killfill: So when you call from Zap and that calls out to SIP you get the AMI message. And when you place a separate call from your SIP device which calls out to your Zap it does not? |
| 14:04.00 | resno | [TK]D-Fender: and i need a sip provider, correct? |
| 14:04.10 | [TK]D-Fender | resno: Nope. |
| 14:04.29 | [TK]D-Fender | resno: you just said you want to use your existing analog line and some phones. |
| 14:04.54 | killfill | [TK]D-Fender: No. Whenever i make a call between the phones. in the same call i expected asterisk to emit ExtensionStatus events about both phones. Its only doing it for the Zap one. |
| 14:05.19 | killfill | whenever i call from the sip of the zap one |
| 14:05.30 | resno | [TK]D-Fender: maybe i said it wrong. i would like to use my existing phones. but use as voip type of setup |
| 14:05.52 | killfill | i can see Dial and Hangup events for the SIP one. but not ExtensionStatus, wich i wish to get.. :) |
| 14:06.14 | [TK]D-Fender | killfill: then you failed to grasp a very basic terminology. Only 1 EXTENSION has been executed |
| 14:06.58 | [TK]D-Fender | [09:59]<resno>[TK]D-Fender: at this moment, only one line. 1 phone. hopefully analog phones <- you just said you have 1 line already |
| 14:06.58 | killfill | hm.. |
| 14:07.11 | [TK]D-Fender | killfill: DEVICES are not PHONES |
| 14:07.19 | [TK]D-Fender | killfill: or EXTENSIONS. |
| 14:07.29 | [TK]D-Fender | killfill: An extension is a line in your dialplan <- |
| 14:07.50 | resno | [TK]D-Fender: ok, correction. i have no current phone. i would like asterisk to handle all that stuff |
| 14:08.06 | [TK]D-Fender | resno: then neither the SPA-3102 ro 3000 is appropriate. |
| 14:08.36 | [TK]D-Fender | resno: Those are for when you want to interface to a line. At this point you want a boring ATA. Go for the Linksys PAP2T-NA |
| 14:09.25 | killfill | [TK]D-Fender: ok, i see. But how does this explain asterisk only sends ExtensionStatus for the (zap) phone? |
| 14:09.53 | [TK]D-Fender | killfill: Zap phone generates the call that STARTS executing extensions in your dialplan |
| 14:09.56 | Pan3D | lol "boring ATA" |
| 14:10.44 | resno | [TK]D-Fender: so this would handle it without any type of funs configuration? |
| 14:11.18 | [TK]D-Fender | resno: What do you want * to do for you? |
| 14:13.28 | killfill | hm.. |
| 14:16.18 | resno | [TK]D-Fender: honestly, i dont honesly know. i kinda want to set it up and see what stuff it can do. |
| 14:17.00 | [TK]D-Fender | resno: well for that you don't need to purchase anything special. You could try it out with just a soft-phone on your PC |
| 14:20.32 | killfill | [TK]D-Fender: i think i still dont get it.. :| |
| 14:22.54 | resno | [TK]D-Fender: im googling how setting up the pap2t-na will be and i dont see any relevant. any help? |
| 14:23.06 | resno | my search term is pap2tna asterisk |
| 14:25.59 | resno | [TK]D-Fender: or does it not even use asterisk? |
| 14:26.18 | hudony | Hi, I need help with multiple extension rigning |
| 14:26.20 | hudony | For some reasons, I can only answer from 1 of the 3 phone rigning |
| 14:26.32 | hudony | the 2 others are cell phones |
| 14:26.34 | hudony | :S |
| 14:28.14 | [TK]D-Fender | resno: there is virtually nothing to set up. |
| 14:28.25 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 14:28.36 | [TK]D-Fender | resno: Its a basic SIP device. You fill in user/pass/IP, and you're pretty much done |
| 14:28.44 | [TK]D-Fender | resno: and it uses whatever you point it to. |
| 14:29.50 | [TK]D-Fender | killfill: You Zap device places a call. That exectues an extension. The End. Dial()-ing a SIP device will not generate an extra event. `A SIP DEVICE is **NOT** and EXTENSION. |
| 14:30.32 | [TK]D-Fender | hudony: pastebin your complete failed attempt from * CLI |
| 14:30.34 | [TK]D-Fender | ~pb |
| 14:30.35 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
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| 14:33.44 | killfill | [TK]D-Fender: ok. So if i want to get a similar behaviour of my sip devices just like the zap extensions, i would have to emit the ExtensionStatus event myself in the dialplan? is it possible to emit such thing in the dialplan? |
| 14:34.37 | [TK]D-Fender | killfill: you are using a completely inappropriate AMI event <- Wrong tool for the wrong job |
| 14:35.06 | [TK]D-Fender | killfill: zap extensions <- WRONG. Zap DEVICE. Stop calling DEVICES as EXTENSIONS. |
| 14:35.18 | killfill | ok sorry.. |
| 14:38.17 | killfill | well im doing a realtime monitor, to see the status of all phones in a web. i noticed i could use StatusExtension event for the Zap device's. For the sip ones, all i see is Dial, hangup and newstate (where state='ringing') event, so i would have to interpret thouse. i thought think would be easier if there were a form to emit the same statueextension events with sip :) |
| 14:40.50 | [TK]D-Fender | killfill: the call does. this has nothing to do with a DEVICE's status however. |
| 14:42.28 | killfill | the call does? what do you mean? |
| 14:44.06 | [TK]D-Fender | killfill: What is is going to take to sink in. ExtensioState is for dialplan lines executing. This has nothing to do with the call having beens tarted by any particular kind of device or that any OTHER device happens to be called because of it. EXTENSION = extensions.conf. You could have a million lines of BS in there for all it cares. |
| 14:44.24 | *** join/#asterisk smooth_penguin (~smoove@59.95.26.94) |
| 14:44.34 | [TK]D-Fender | killfill: it is not a "phone status" tool. Whong tool. Are you following me? |
| 14:44.42 | [TK]D-Fender | wrong* |
| 14:47.15 | killfill | ExtensionStatus it tells me information about the zap devices i wish to show. But i think i follow you. i should not base this on that event. |
| 14:48.21 | killfill | (that is what you mean right?...) |
| 14:48.48 | [TK]D-Fender | killfill: No, it shows you *&#$ing extensions.conf executing. |
| 14:50.01 | devoid | someone is angry |
| 14:50.41 | [TK]D-Fender | devoid: 3/4 of an hour an someone is not getting the picture. |
| 14:50.56 | killfill | yes, but you mean i should not use AMI/AJAM for this?.. |
| 14:51.04 | DennisG | just read the Asterisk ebook ? |
| 14:51.07 | killfill | heh.. yes, maybe i became blind on this |
| 14:52.00 | [TK]D-Fender | killfill: ExtensionState is the WRONG FUCKING EVENT TYPE TO LOOK AT. I did not say AMI did not hanve something to do what you want. I said that you picked the WRONG FUCKING EVENT. ExtensionState has JACK SHIT to do with what you want! |
| 14:52.23 | killfill | oh yes, i got that. |
| 14:52.28 | [TK]D-Fender | killfill: No go read the event list to see what DOES cover DEVICES |
| 14:52.37 | [TK]D-Fender | now8 |
| 14:54.08 | killfill | sure, i think they are the call status events, like dial, hangup, etc. i was using too.. :) |
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| 14:56.07 | |amadeus| | *lol* |
| 14:57.54 | |amadeus| | so did you guys ever thought about talking via musiconhold? |
| 14:58.13 | |amadeus| | Im actually going to implement VoiceOverRingtone() |
| 15:01.51 | [TK]D-Fender | |amadeus|: Doable.... through more mundane means I suppose as well... use a streaming source for MoH. Have that running off the line-in on a sound card. On that same system set up a softphone with auto-answer on it. You can then talk over the music |
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| 15:05.20 | |amadeus| | sounds not like a perfect solution but might work for me ... I was wondering if the mic is turned on during the invite on my mobile phone? Tested it with Echo() but didn't work for me |
| 15:06.45 | [TK]D-Fender | |amadeus|: I don't see a call to debug or any hint of how you're calling it... |
| 15:07.00 | *** part/#asterisk resno (~bryan@cpe-098-026-005-130.nc.res.rr.com) |
| 15:09.11 | |amadeus| | yeah Im just saying ... had an Echo() in the dailplan and called it with my mobile but wasn't able to hear my voice |
| 15:09.28 | |amadeus| | spent really not much time on it to debug, will do that |
| 15:09.31 | [TK]D-Fender | |amadeus|: And I'm just saying we have no details about how this call arrives at * in the first place... |
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| 15:11.15 | |amadeus| | ok here it comes, I've got an actual phone number linked to my asterisk .. so I was calling that number from my mobile and just did an Echo() in the dailplan (without answering it) |
| 15:11.41 | [TK]D-Fender | |amadeus|: ..... |
| 15:11.53 | [TK]D-Fender | |amadeus|: Just not sinking in is it? |
| 15:13.10 | |amadeus| | I know it must be my bad |
| 15:13.25 | [TK]D-Fender | |amadeus|: how does this call ARRIVE AT ASTERISK? |
| 15:14.07 | |amadeus| | from another asterisk, my asterisk is just a client in this case |
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| 15:17.15 | [TK]D-Fender | goes off to do something productive... |
| 15:17.40 | |amadeus| | ok must be my English then |
| 15:17.52 | [TK]D-Fender | no |
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| 15:23.03 | |amadeus| | anyway, maybe there's a chance to bridge these 2 channels together |
| 15:25.21 | |amadeus| | just wants short free phone calls from mobiles |
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